mirror of
https://github.com/mpv-player/mpv
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d7f6cb23de
Substract the delay caused by filter buffering when calculating currently playing audio position. This matters for af_scaletempo which buffers significant and varying amounts of data. For other current filters the effect is normally insignificant. Instead of the old time-based filter delay field (which was ignored) this version stores the per-filter delay in units of bytes input read without corresponding output. This allows the current scaletempo behavior where other filters before and after it can see the same nominal samplerate even though the real duration of the data varies; in this case the other filters can not know the delay they're causing in terms of real time. git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@24928 b3059339-0415-0410-9bf9-f77b7e298cf2
249 lines
6.3 KiB
C
249 lines
6.3 KiB
C
/*=============================================================================
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//
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// This software has been released under the terms of the GNU General Public
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// license. See http://www.gnu.org/copyleft/gpl.html for details.
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//
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// Copyright 2001 Anders Johansson ajh@atri.curtin.edu.au
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//
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//=============================================================================
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*/
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/* Equalizer filter, implementation of a 10 band time domain graphic
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equalizer using IIR filters. The IIR filters are implemented using a
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Direct Form II approach, but has been modified (b1 == 0 always) to
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save computation.
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*/
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#include <stdio.h>
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#include <stdlib.h>
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#include <inttypes.h>
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#include <math.h>
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#include "af.h"
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#define L 2 // Storage for filter taps
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#define KM 10 // Max number of bands
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#define Q 1.2247449 /* Q value for band-pass filters 1.2247=(3/2)^(1/2)
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gives 4dB suppression @ Fc*2 and Fc/2 */
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/* Center frequencies for band-pass filters
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The different frequency bands are:
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nr. center frequency
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0 31.25 Hz
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1 62.50 Hz
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2 125.0 Hz
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3 250.0 Hz
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4 500.0 Hz
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5 1.000 kHz
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6 2.000 kHz
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7 4.000 kHz
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8 8.000 kHz
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9 16.00 kHz
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*/
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#define CF {31.25,62.5,125,250,500,1000,2000,4000,8000,16000}
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// Maximum and minimum gain for the bands
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#define G_MAX +12.0
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#define G_MIN -12.0
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// Data for specific instances of this filter
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typedef struct af_equalizer_s
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{
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float a[KM][L]; // A weights
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float b[KM][L]; // B weights
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float wq[AF_NCH][KM][L]; // Circular buffer for W data
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float g[AF_NCH][KM]; // Gain factor for each channel and band
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int K; // Number of used eq bands
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int channels; // Number of channels
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float gain_factor; // applied at output to avoid clipping
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} af_equalizer_t;
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// 2nd order Band-pass Filter design
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static void bp2(float* a, float* b, float fc, float q){
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double th= 2.0 * M_PI * fc;
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double C = (1.0 - tan(th*q/2.0))/(1.0 + tan(th*q/2.0));
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a[0] = (1.0 + C) * cos(th);
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a[1] = -1 * C;
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b[0] = (1.0 - C)/2.0;
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b[1] = -1.0050;
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}
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// Initialization and runtime control
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static int control(struct af_instance_s* af, int cmd, void* arg)
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{
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af_equalizer_t* s = (af_equalizer_t*)af->setup;
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switch(cmd){
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case AF_CONTROL_REINIT:{
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int k =0, i =0;
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float F[KM] = CF;
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s->gain_factor=0.0;
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// Sanity check
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if(!arg) return AF_ERROR;
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af->data->rate = ((af_data_t*)arg)->rate;
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af->data->nch = ((af_data_t*)arg)->nch;
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af->data->format = AF_FORMAT_FLOAT_NE;
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af->data->bps = 4;
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// Calculate number of active filters
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s->K=KM;
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while(F[s->K-1] > (float)af->data->rate/2.2)
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s->K--;
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if(s->K != KM)
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af_msg(AF_MSG_INFO,"[equalizer] Limiting the number of filters to"
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" %i due to low sample rate.\n",s->K);
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// Generate filter taps
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for(k=0;k<s->K;k++)
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bp2(s->a[k],s->b[k],F[k]/((float)af->data->rate),Q);
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// Calculate how much this plugin adds to the overall time delay
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af->delay = 2 * af->data->nch * af->data->bps;
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// Calculate gain factor to prevent clipping at output
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for(k=0;k<AF_NCH;k++)
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{
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for(i=0;i<KM;i++)
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{
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if(s->gain_factor < s->g[k][i]) s->gain_factor=s->g[k][i];
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}
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}
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s->gain_factor=log10(s->gain_factor + 1.0) * 20.0;
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if(s->gain_factor > 0.0)
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{
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s->gain_factor=0.1+(s->gain_factor/12.0);
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}else{
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s->gain_factor=1;
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}
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return af_test_output(af,arg);
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}
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case AF_CONTROL_COMMAND_LINE:{
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float g[10]={0.0,0.0,0.0,0.0,0.0,0.0,0.0,0.0,0.0,0.0};
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int i,j;
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sscanf((char*)arg,"%f:%f:%f:%f:%f:%f:%f:%f:%f:%f", &g[0], &g[1],
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&g[2], &g[3], &g[4], &g[5], &g[6], &g[7], &g[8] ,&g[9]);
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for(i=0;i<AF_NCH;i++){
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for(j=0;j<KM;j++){
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((af_equalizer_t*)af->setup)->g[i][j] =
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pow(10.0,clamp(g[j],G_MIN,G_MAX)/20.0)-1.0;
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}
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}
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return AF_OK;
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}
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case AF_CONTROL_EQUALIZER_GAIN | AF_CONTROL_SET:{
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float* gain = ((af_control_ext_t*)arg)->arg;
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int ch = ((af_control_ext_t*)arg)->ch;
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int k;
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if(ch >= AF_NCH || ch < 0)
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return AF_ERROR;
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for(k = 0 ; k<KM ; k++)
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s->g[ch][k] = pow(10.0,clamp(gain[k],G_MIN,G_MAX)/20.0)-1.0;
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return AF_OK;
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}
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case AF_CONTROL_EQUALIZER_GAIN | AF_CONTROL_GET:{
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float* gain = ((af_control_ext_t*)arg)->arg;
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int ch = ((af_control_ext_t*)arg)->ch;
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int k;
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if(ch >= AF_NCH || ch < 0)
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return AF_ERROR;
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for(k = 0 ; k<KM ; k++)
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gain[k] = log10(s->g[ch][k]+1.0) * 20.0;
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return AF_OK;
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}
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}
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return AF_UNKNOWN;
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}
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// Deallocate memory
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static void uninit(struct af_instance_s* af)
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{
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if(af->data)
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free(af->data);
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if(af->setup)
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free(af->setup);
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}
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// Filter data through filter
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static af_data_t* play(struct af_instance_s* af, af_data_t* data)
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{
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af_data_t* c = data; // Current working data
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af_equalizer_t* s = (af_equalizer_t*)af->setup; // Setup
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uint32_t ci = af->data->nch; // Index for channels
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uint32_t nch = af->data->nch; // Number of channels
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while(ci--){
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float* g = s->g[ci]; // Gain factor
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float* in = ((float*)c->audio)+ci;
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float* out = ((float*)c->audio)+ci;
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float* end = in + c->len/4; // Block loop end
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while(in < end){
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register int k = 0; // Frequency band index
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register float yt = *in; // Current input sample
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in+=nch;
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// Run the filters
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for(;k<s->K;k++){
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// Pointer to circular buffer wq
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register float* wq = s->wq[ci][k];
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// Calculate output from AR part of current filter
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register float w=yt*s->b[k][0] + wq[0]*s->a[k][0] + wq[1]*s->a[k][1];
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// Calculate output form MA part of current filter
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yt+=(w + wq[1]*s->b[k][1])*g[k];
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// Update circular buffer
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wq[1] = wq[0];
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wq[0] = w;
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}
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// Calculate output
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*out=yt*s->gain_factor;
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out+=nch;
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}
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}
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return c;
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}
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// Allocate memory and set function pointers
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static int af_open(af_instance_t* af){
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af->control=control;
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af->uninit=uninit;
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af->play=play;
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af->mul=1;
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af->data=calloc(1,sizeof(af_data_t));
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af->setup=calloc(1,sizeof(af_equalizer_t));
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if(af->data == NULL || af->setup == NULL)
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return AF_ERROR;
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return AF_OK;
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}
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// Description of this filter
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af_info_t af_info_equalizer = {
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"Equalizer audio filter",
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"equalizer",
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"Anders",
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"",
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AF_FLAGS_NOT_REENTRANT,
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af_open
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};
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